20 Commits

Author SHA1 Message Date
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
Artem Titov
1ebfb6aac7 Introduce VideoFrame::id to keep track of frames inside application.
Also switch webrtc code from deprecated constructors to the builder API.

Change-Id: Ie325bf1e9b4ff1e413fef3431ced8ed9ff725107
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/114422
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26132}
2019-01-04 08:59:26 +00:00
Niels Möller
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
Niels Möller
4dc66c53d0 Move EncodedImage class to api/video/
Bug: webrtc:9378
Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba
Reviewed-on: https://webrtc-review.googlesource.com/c/104002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-08 07:37:10 +00:00
Niels Möller
f5033ad201 In (new) estimator of encoder cpu load, count max per input frame.
Bug: webrtc:9619
Change-Id: Ifc874fa3bd069e48a10fec57b673546aafd070e3
Reviewed-on: https://webrtc-review.googlesource.com/94143
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24293}
2018-08-15 15:16:47 +00:00
Niels Möller
58d2a5e976 Tolerate out of order samples to SendProcessingUsage2::FrameSent.
Bug: chromium:842613
Change-Id: I57e4df75dcfdfb9bf42819f31d2186e875a90a3a
Reviewed-on: https://webrtc-review.googlesource.com/92880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24224}
2018-08-08 09:58:42 +00:00
Niels Möller
213618e37e New api function CreateVideoStreamEncoder.
Bug: webrtc:8830
Change-Id: I01de86f601e48f76e6b41b4182ce006d397a190c
Reviewed-on: https://webrtc-review.googlesource.com/78260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24079}
2018-07-24 09:14:26 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Danil Chapovalov
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00
Niels Möller
d1f7eb6e83 Postpone setting of CpuOveruseOptions.
This will enable changing thresholds when switching between hardware
and software encoders. It is also a partial revert of
https://webrtc-review.googlesource.com/33340: construction of the
OveruseFrameDetector is still in VideoSendStream, but configuration is
moved back to VideoStreamEncoder.

Longer term, information about HW vs SW, or generally, about resources
consumed by the encoder, should be passed in the per-frame callbacks
to OveruseFrameDetector, and then the CpuOveruseOptions could move
back to construction time.

Bug: webrtc:8504, webrtc:8830
Change-Id: I44577519d4e05356730cac9bd9ae3c74bfc17ed7
Reviewed-on: https://webrtc-review.googlesource.com/65163
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22761}
2018-04-06 08:34:32 +00:00
Niels Möller
73f29cbcc1 Move creation of OveruseFrameDetector to VideoSendStream.
Intended to make it easier to wire up cpu-adaptation experiments.
To setup the circular references between OveruseFrameDetector and
VideoStreamEncoder, let the AdaptationObserverInterface pointer be
an argument to StartCheckForOveruse.

Bug: webrtc:8504
Change-Id: Ifcf7655ec65e637819d77f507552cb22a6aa5f0f
Reviewed-on: https://webrtc-review.googlesource.com/33340
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22006}
2018-02-13 15:45:03 +00:00
Niels Möller
83dbeacb1a Add alternative load estimator to OverUseFrameDetector.
The new estimator uses the timestamps attached to EncodedImage, and is
taken from the reverted cl
https://webrtc-review.googlesource.com/c/src/+/23720.

Bug: webrtc:8504
Change-Id: I273bbe3eb6ea2ab9628c9615b803a379061ad44a
Reviewed-on: https://webrtc-review.googlesource.com/31380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21289}
2017-12-15 10:10:06 +00:00
Niels Möller
e541be78f9 Add OveruseFrameDetector tests with random inter-frame intervals
This is a reland of the tests added in the reverted cl
https://webrtc-review.googlesource.com/c/src/+/23720, with
expectations relaxed to make tests pass also with the current (old)
estimator.

Bug: webrtc:8504
Change-Id: I69fd8cc7e87e05b24be75b146f1cac91c5f96f46
Reviewed-on: https://webrtc-review.googlesource.com/30142
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21243}
2017-12-13 13:10:11 +00:00
Niels Möller
6b642f730c Delete EncodedFrameObserver::OnEncodeTiming.
This callback was used only by the PrintSamplesToFile feature of
video_quality_test, which looks like it has been broken for some time
(due to mixup of capture time and ntp time).

Bug: webrtc:8504
Change-Id: I7d2b55405caeffda582ae0d6fb0e7dfdfce4c5a9
Reviewed-on: https://webrtc-review.googlesource.com/31420
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21211}
2017-12-11 16:41:46 +00:00
Niels Möller
7dc26b7b32 Revert "Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage."
This reverts commit eee7cedf6901ad5616dbf0bf09f35010207f823d.

Reason for revert: Intend to rework with a flag to select between old or new estimator, to be wired up as an origin-trial experiment. And old estimator deleted at a later point.

Original change's description:
> Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage.
> 
> Bug: webrtc:8504
> Change-Id: I3f99c3c1e528f59b45724921a91f65b485f5b820
> Reviewed-on: https://webrtc-review.googlesource.com/23720
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20979}

TBR=nisse@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8504
Change-Id: Ieec4624e1d6dd8472b7e89c7bd19f425d9b54533
Reviewed-on: https://webrtc-review.googlesource.com/30180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21115}
2017-12-06 10:32:12 +00:00
Niels Möller
eee7cedf69 Refactor OverUseFrameDetector to use the timestamps attached to EncodedImage.
Bug: webrtc:8504
Change-Id: I3f99c3c1e528f59b45724921a91f65b485f5b820
Reviewed-on: https://webrtc-review.googlesource.com/23720
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20979}
2017-12-01 15:13:26 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00